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Release Notes - NFS Professional Services

1. Supported SIP Headers Header Field Accept Accept Encoding Alert Info Allow Also Asserted ldentity Authorization Call ID Call Info Contact Content Disposition Content Encoding Content Length Content Type Cseq Date Diversion Encryption Version 5 6 35 Supported Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes No 2 Supported Features September 2008 ca AudioCodes Header Field Expires Fax From History Info Join Max Forwards Messages Waiting MIN SE Organization P Associated URI P Asserted Identity P Charging Vector P Preferred Identity Priority Proxy Authenticate Proxy Authorization Proxy Require Prack Reason Record Route Refer To Referred By Replaces Require Remote Party ID Response Key Retry After Route Rseq Session Expires Server Service Route SIP If Match Subject Supported SIP Release Notes 36 Supported Yes Yes Yes Yes Yes Yes Yes Yes No Yes Receive Only Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes MediaPack Series Document LTRT 65611 SIP Release Notes Header Field Target Dialog Timestamp To Unsupported User Agent Via Voicemail Warning WWW Authenticate 2 2 4 SDP Headers The device supports the following SDP Headers Supported Yes Yes Yes Yes Yes Yes Yes Yes Yes Table 2 4 Supported SDP Headers SDP
2. the Web interface does not exit the Scenario mode On the Software Upgrade Wizard page the software upgrade process must be completed prior to clicking the Back button Clicking the Back button before the wizard completes causes a display distortion The following pages cannot be added to a Scenario e Web User Accounts e Web amp Telnet Access List e Regional Settings For users who have Read Only access to the Web interface the Read Only Mode string text does not appear in bold format on the following pages Tel to IP Routing Table SNMP Community String and SNMP Trap Destinations The IP Routing Table page can be configured in the Web interface however the ini file is not updated with the new settings Not all parameters can be changed on the fly in the Web interface Parameters that can t be changed on the fly are depicted with the lightning amp symbol To change these parameters reset the device using the Web interface s Reset button When changing device parameters in the Web interface the new parameters are permanently stored in flash memory only after the device is reset from the Web or after the BURN button is clicked in the Maintenance Actions page The number of fax calls displayed in the fields Attempted Fax Calls Counter and Successful Fax Calls Counter in the Calls Count pages may not be accurate In the Coders and Coder Group Settings pages the voice quality
3. For example Dns2lp Dns2lp 0 DnsName 1 1 1 1 2 2 2 2 3 3 3 3 4 4 4 4 Dns2lp Notes This parameter can include up to 20 indices lf the internal DNS table is used the device first attempts to resolve a domain name using this table If the domain name isn t found the device performs a DNS resolution using an external DNS server Modification Description and Web interface reference Determines the index of the first Ringback Tone in the CPT file This option enables an Application server to request the device to play a distinctive Ringback tone to the calling party according to the destination of the call The tone is played according to the Alert Info header received in the 180 Ringing SIP response the value of the Alert Info header is added to the value of this parameter The valid range is 1 to 1 000 The default value is 1 i e play standard Ringback tone Notes It is assumed that all Ringback Tones are defined in sequence in the CPT file Incase of an MLPP call the device uses the value of this parameter plus one as the index of the Ringback tone in the CPT file e g if this value is set to 1 then the index is 2 i e 1 1 25 September 2008 7a tal AudioCodes MediaPack Series Parameter Description PeerHostNameVerificationM Modification Support for the asterisk wildcard ode Determines whether the device verifies the Subject Name of a remote certificate wh
4. Header Element v Protocol version o Owner creator and session identifier a Attribute information c Connection information d Digit m Media name and transport address s Session information t Time alive header b Bandwidth header u Uri Description Header e Email Address header i Session Info Header p Phone number header y Year Version 5 6 of 2 Supported Features Supported Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes September 2008 7a e AudioCodes MediaPack Series 2 2 5 SIP Responses The device supports the following SIP responses m 1xx Response Information Responses m 2xx Response Successful Responses m 3xx Response Redirection Responses m 4xx Response Client Failure Responses m 5xx Response Server Failure Responses m 6xx Response Global Responses 2 2 5 1 1xx Response Information Responses Table 2 5 Supported 1xx SIP Responses 1xx Response Supported Comments 100 Trying Yes The SIP device generates this response upon receiving a Proceeding message from ISDN or immediately after placing a call for CAS signaling 180 Ringing Yes The SIP device generates this response for an incoming INVITE message Upon receiving this response the device waits for a 200 OK response 181 Call is Yes The SIP device doesn t generate these responses However the Being device does receive them The device processes these Forwarded respons
5. according to the user part of the URI only Otherwise the binding is according to the entire URI i e User Host default 1 User Part only The binding is always performed according to the User Part only Determines whether the SAS application uses ENUM queries to route incoming INVITE requests when in Emergency mode Once an INVITE is received in Emergency mode the SAS database of registered users is searched for a matching AoR If not found the Redundant SAS servers are searched If there is still no match an ENUM query is performed and the response is used to correctly route the INVITE If no response is received from the ENUM server the INVITE is routed to the default gateway 0 Disable default 1 Enable This ini file table parameter is used by the SAS application to manipulate the User Part of an incoming REGISTER request AoR the To header before saving it to the registered users database The format of this table parameter is as follows SASRegistrationManipulation FORMAT SASRegistrationManipulation_Index SASRegistrationManipulation_ RemoveFromRight SASRegistrationManipulation_LeaveFromRight SASRegistrationManipulation RemoveFromRight number of digits removed from the right side of the User Part before saving to the registered user database 20 Document LTRT 65611 SIP Release Notes Parameter SIP Rerouting Mode SIPReroutingMode Master Key Identifier MKI Size SRTPT
6. from Normal mode to Emergency mode This alarm is cleared once the SAS returns to Normal mode 2 SNMP Actions for X 509 Certificates MP 124 MP 11x FXS FXO The following SNMP actions were added for X 509 certificates e acSysSecurityGenCsrSubjectName Generates a certificate signing request using the provided name e acSysSecuritySelfSignedCertificateSubjectName Generates a Self Signed Certificate using the provided name SIP Release Notes 18 Document LTRT 65611 SIP Release Notes 1 6 New Parameters 1 What s New in Release 5 6 The table below describes the new parameters for Release 5 6 Most of these new parameters can be configured using both the ini file enclosed in square brackets and the Web interface Table 1 1 Release 5 6 New Web ini File Parameters Parameter Held Timeout HeldTimeout EnableComfortTone XferPrefixIP2Tel Alt Routing Tone Duration AltRoutingToneDuration SAS Proxy Set SASProxySet Redundant SAS Proxy Set RedundantSASProxySet Version 5 6 Description Determines the time interval that the device can allow a call to remain on hold If a Resume un hold Re INVITE message is received before the timer expires the call is renewed If this timer expires the call is released 1 The call is placed on hold indefinitely until the initiator of on hold retrieves the call again default 0 2400 Time to wait in seconds after which t
7. has been received the device responds with an ACK and disconnects the call 39 September 2008 ca AudioCodes 4xx Response 407 Proxy Authentication Required 408 Request Timeout 409 Conflict 410 Gone 411 Length Required 413 Request Entity Too Large 415 Unsupported Media 420 Bad Extension 423 Interval Too Brief 433 Anonymity Disallowed 480 Temporarily Unavailable 481 Call Leg Transaction Does Not Exist 482 Loop Detected 483 Too Many Hops SIP Release Notes Supported Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes MediaPack Series Comments Authentication support for Basic and Digest Upon receiving this message the device issues a new request according to the scheme received on this response The device generates this response if the no answer timer expires Upon receipt of this message and before a 200 OK has been received the device responds with an ACK and disconnects the call The device doesn t generate this response Upon receipt of this message and before a 200 OK has been received the device responds with an ACK and disconnects the call The device doesn t generate this response Upon receipt of this message and before a 200 OK has been received the device responds with an ACK and disconnects the call The device doesn t generate this response Upon receipt of this message and before a 200 OK has been received the devic
8. then resends the request to a new address 2 2 5 4 4xx Response Client Failure Responses 4xx Response 400 Bad Request 401 Unauthorized 402 Payment Required 403 Forbidden 404 Not Found 405 Method Not Allowed 406 Not Acceptable Version 5 6 Table 2 8 Supported 4xx SIP Responses Supported Yes Yes Yes Yes Yes Yes Yes Comments The device doesn t generate this response Upon receipt of this message and before a 200 OK has been received the device responds with an ACK and disconnects the call Authentication support for Basic and Digest Upon receiving this message the device issues a new request according to the scheme received on this response The device doesn t generate this response Upon receipt of this message and before a 200 OK has been received the device responds with an ACK and disconnects the call The device doesn t generate this response Upon receipt of this message and before a 200 OK has been received the device responds with an ACK and disconnects the call The SIP device generates this response if it is unable to locate the callee Upon receiving this response the device notifies the User with a Reorder Tone The device doesn t generate this response Upon receipt of this message and before a 200 OK has been received the device responds with an ACK and disconnects the call The device doesn t generate this response Upon receipt of this message and before a 200 OK
9. to Supported header Supports RFC 3581 Symmetric Response Routing Supports RFC 3605 RTCP Attribute in SDP Supports RFC 3326 Reason header Supports RFC 4028 Session Timers in SIP Supports network asserted identity and privacy RFC 3325 and RFC 3323 Support RFC 3903 SIP Extension for Event State Publication Support RFC 3953 The Early Disposition Type for SIP Support for RFC 3966 The tel URI for Telephone Numbers Support RFC 4244 An Extension to SIP for Request History Information Supports Tel URI Uniform Resource Identifier according to RFC 2806 bis SIP Release Notes 32 Document LTRT 65611 SIP Release Notes 2 Supported Features Version 5 6 Supports ITU V 152 Procedures for supporting Voice Band Data over IP Networks Remote party ID lt draft ietf sip privacy 04 txt gt Supports obtaining Proxy Domain Name s from DHCP Dynamic Host Control Protocol according to RFC 3361 Supports handling forking proxy multiple responses RFC 2833 Relay for DTMF Digits including payload type negotiation DTMF out of band transfer using e INFO method lt draft choudhuri sip info digit 00 txt gt e INFO method compatible with Cisco gateways e NOTIFY method lt draft mahy sipping signaled digits 01 txt gt e INFO method compatible with Korea Telecom format SIP URL sip phone number IP address such as 1225556 10 1 2 4 where 122556 is the phone number of the source or destination or sip
10. Ces AudioCodes CPE amp Access Analog Gateways SIP MediaPack MP 124 amp MP 11x Release Notes Version 5 6 Document LTRT 65611 September 2008 SIP Release Notes Contents Table of Contents 1 i 4 bee Prodi 1 1 2 Support 14 Ha t 1 4 Web New Features 1 5 SNMP New Features i Pee al ag uy 2 ene en oe eee ROE Sees ee ner eRe iO Mer Our NOMEN rre mene 17 Modified Picea etree ai sise a deas air eerdeeed ame 1 8 Obsolete Panait E E EA E A A E E T E A 2 Supported Fea eS ann 31 2 1 SIP E 3 1 3 2 35 CLI Casha 4 Resolved ConstrainiS sisssssirirsscevsisscuurousirurusisvunodsivuousonisuocukuinnsevurinssversisnoassi AF A web NT e e easa A A A E A N A T 5 Earlier Releases COPE 49 Version 5 6 3 September 2008 7a T 4 id AudioCodes MediaPack Series List of Figures Figure 1 1 Double Hod SIP Gall Pi siioni ia aaie aaa ia 9 Foute 1 2 Gascading Am EXAMP mmmmuwwvwwvwwuvwevwwvvvwwewwwvvvewwvvwwwwwwvvwwvwwevvvvvvwvvvvvv 15 Fours tsk Fowchan o SAS POES mm mm m m u uwewvwvwwvweuvwwwvuvwvvwvvvvwvvvevvwvvvwvwvvwvwvvwvvv 16 Figure 1 4 ENUM Support for SAS ApplhicatiOl sisikii a aa 17 List of Tables Table 1 1 Release 5 6 New Web ini File ParameterS cccssceseseeceeeceeeeeeeneeeeeeeeseeeeeeaeseeneeenaes 19 Table 1 2 Release 5 6 Modified Web ini File Parameters 0 ccccccesccecceeeeeeeeeeeeeeeeeeeeaeseeaeeenaes 23 Table 1 3 Releas
11. FromLeft Number of stripped digits from the left RemoveFromRight Number of stripped digits from the right LeaveFromRight Number of remaining digits from the right Prefix2Add String to add as prefix Suffix2Add String to add as suffix IsPresentationRestricted N A set to SrcTrunkGroupID Source Trunk Group ID SrcIPGrouplD Source IP Group ID For example NumberMapTel2Ip NumberMapTel2Ip 0 01 0 0 2 971 NumberMapTel2lp 1 10 10 255 255 3 0 5 100 255 NumberMapTel2Ip Notes RemoveFromLeft RemoveFromRight Prefix2Add Suffix2Add LeaveFromRight NumberType and NumberPlan are applied if the called and calling numbers match the DestinationPrefix and SourcePrefix conditions The manipulation rules are executed in the following order RemoveFromLeft RemoveFromRight LeaveFromRight Prefix2Add and Suffix2Add Parameters can be skipped by using two dollar signs Number Plan and Type can optionally be used in Remote Party ID RPID header by using the EnableRP Header and AddTON2RPI parameters Modification New parameters for Source Trunk Group and Source IP Group This ini file table parameter manipulates the source phone number for Tel to IP calls The format of this parameter is as follows SourceNumberMapTel2 lp FORMAT SourceNumberMapTel2lp_Index SourceNumberMapTel2lp_DestinationPrefix SourceNumberMapTel2lp_SourcePre
12. Generates and uses a Master Key Identifier MKI value on outgoing SRTP streams in addition to existing support for incoming SRTP streams e Supports SRTP SRTCP attributes as defined in RFC 4568 SDP Security Descriptions for Media Streams UNAUTHENTICATED_SRTP UNENCRYPTED_SRTCP and UNENCRYPTED_SRTP Relevant parameters SRTPTxPacketMkK ISize RTPAuthenticationDisable Tx RTPEncryptionDisableTx RTCPEncryptionDisableTx 12 MLPP Enhancements MP 124 Version 5 6 The device s support for the Multi Level Precedence and Preemption MLPP protocol has been enhanced to support Supplementary Services scenarios such as e Call Hold e Call Transfer e Call Waiting e 3 Way Conference using an external Media Server For a detailed description of the MLPP implementation using SIP please refer to the device s User s Manual 13 Distinctive Ringback Tones MP 11x FXS FXO The device can now play a specific Ringback Tone defined in the Call Progress Tones file This option enables an Application server to request the device to play a distinctive Ringback tone to the calling party according to the destination of the call The tone is played according to the Alert Info header received in the 180 Ringing SIP response Relevant parameter FirstCall RBTId 11 September 2008 7a T tal AudioCodes MediaPack Series 14 Play Tone upon Alternative Routing The device can now play a tone whenever Alternative Routing is used Eac
13. S IP address appears in the SIP via header of the request it is not forwarded this prevents loops in the request s course If no such redundant SAS exists the SAS sends the request to its default gateway defined by the parameter SASDefaultGatewayIP Figure 1 2 Cascading SAS Example IP PBX Version 5 6 15 September 2008 7a T id AudioCodes MediaPack Series Figure 1 3 Flowchart of SAS Process Move to the Next Redundant SAS If all Redundant SAS are Offline send Request to Default GW SIP Release Notes 16 Document LTRT 65611 SIP Release Notes 1 What s New in Release 5 6 3 ENUM Support for SAS Application MP 124 MP 11x FXS FXO A new option was added to the SAS application allowing Telephone Number Mapping ENUM E 164 capabilities to route incoming INVITE requests Once an INVITE is received in Emergency mode the SAS database of registered users is searched for a matching Address Of Record AoR If not found the Redundant SAS servers are searched If there is still no match an ENUM query is performed and the response is used to correctly route the INVITE Figure 1 4 ENUM Support for SAS Application Receiving INVITE request Check the SAS DB for intemal registered users Check for online Send INVITE redundant SAS server YES to registered Via limitations user Send INVITE ple NO Sena YES to redundant ENUM query server SAS Send INVITE AR GAY NO YES according to ENUM answ
14. al 33 September 2008 7a 4 id AudioCodes MediaPack Series uW Support RFC 4235 An INVITE Initiated Dialog Event Package for SIP Partial m Support RFC 3680 A SIP Event Package for Registrations 2 1 2 Unsupported SIP Features The following SIP features are not supported m MESSAGE method m Preconditions RFC 3312 m SDP Simple Capability Declaration RFC 3407 m S MIME 2 2 SIP Compliance Tables The SIP device complies with RFC 3261 as shown in the following subsections 22 1 SIP Functions The device supports the following SIP Functions Table 2 1 Supported SIP Functions Function Supported User Agent Client UAC Yes User Agent Server UAS Yes Proxy Server Third party only tested with amongst others Ubiquity Delta3 Microsoft 3Com BroadSoft Snom and Cisco Proxies Redirect Server Third party Registrar Server Third party Event Publication Agent Yes EPA Event State Compositor Third party ESC 2 2 2 SIP Methods The device supports the following SIP Methods Table 2 2 Supported SIP Methods Method Supported Comments INVITE Yes ACK Yes BYE Yes SIP Release Notes 34 Document LTRT 65611 SIP Release Notes Method Supported CANCEL Yes REGISTER Yes Send only REFER Yes NOTIFY Yes INFO Yes OPTIONS Yes PRACK Yes UPDATE Yes PUBLISH Yes Send only SUBSCRIBE Yes 22 3 SIP Headers The device supports the following SIP Headers Comments Inside and outside of a dialog Table 2 3
15. an IPSec configuration table are incorrect The user should override the default values before activating the new row The acBoardConfigurationError alarm trap generated as a result of a configuration error does not clear The following RTP MIB objects are not supported rtpRcevrSRCSSRC rtpRevrSSRC rtpSenderSSRC rtpRevrLostPackets rtpRcvrPackets rtpSenderPackets rtpRcvrOctets and rtpSenderOctets An Ethernet link trap is sent before the link is up manager does not receive clear This occurs because a spanning tree algorithm is being calculated in the Ethernet switch The following encryptions types are currently supported for SNMP v3 users only e Authentication protocol MD5 and SHA e Privacy protocol DES and AES128 The range of the faxModemRelayVolume MIB object is incorrect Instead of 0 to 15 it should be 18 to 3 corresponding to an actual volume of 18 5 dBm to 3 5 dBm Only one SNMP manager can access the device simultaneously 3 5 CLI Constraints This release includes the following known command line interface CLI constraint 1 When connecting to a device using Telnet CLI Syslog messages do not appear by default The show log command must be used to enable this feature SIP Release Notes 46 Document LTRT 65611 SIP Release Notes 4 Resolved Constraints 4 Resolved Constraints 4 1 Web Interface The following Web interface constraints from previous releases have now been resolved in Relea
16. ble The registration request is resent according to the parameter RegistrationTimeDivider For example if RegistrationTimeDivider 70 and Registration Expires time 3600 the device resends its registration request after 3600 x 70 2520 sec The default value of RegistrationTimeDivider is 50 If registration per channel is selected on device startup the device sends REGISTER requests according to the maximum number of allowed SIP dialogs configured by the parameter NumberOfActiveDialogs After each received response the subsequent REGISTER request is sent m Proxy and Registrar Authentication handling 401 and 407 responses using Digest method Accepted challenges are kept for future requests to reduce the network traffic W Single device Registration or multiple Registration of all device endpoints Supported methods INVITE CANCEL BYE ACK REGISTER OPTIONS INFO REFER UPDATE NOTIFY PRACK SUBSCRIBE and PUBLISH Modifying connection parameters for an already established call re INVITE Working with Redirect server and handling 3xx responses Early media supporting 183 Session Progress PRACK reliable provisional responses RFC 3262 Call Hold and Transfer Supplementary services using REFER Refer To Referred By Replaces and NOTIFY messages Supports RFC 3711 Secured RTP and Key Exchange according to RFC 4568 Supports RFC 3489 Simple Traversal of UDP Through NATs STUN Supports RFC 3327 Adding Path
17. cation but C D and the FSK are generated by the supplied service The signal can be generated using the UDT signal generation mechanism e Prior to the detection of NTT Caller ID Type 2 two DTMF detections C and D remain unscreened Setting the V 21 Transport Type to Bypass and the Fax Transport Type to Relay results in entering the Fax Relay mode at the 2 100 Hz signal Only at the end of this signal does the channel enter Bypass mode Transparent With Events Bell modem Transport Type is not supported The RFC 2198 redundancy mode with RFC 2833 is not supported i e if a complete DTMF digit is lost it is not reconstructed The current RFC 2833 implementation supports redundancy for inter digit information lost 43 September 2008 7a 4 id AudioCodes MediaPack Series T 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 The resolution of the duration of digits On and Off time when dialing to the IP side using RFC 2833 relay is dependent on the basic frame size of the coder being used Incoming CNG T 38 packets do not switch the channel to T 38 mode When the fax CNG detector is not Transparent a fax CNG tone received from the TDM cannot be detected using the Call Progress Tone detector Debug Recording e Only one IP target is allowed e Maximum of 50 trace rules are allowed simultaneously e Maximum of 5 media stream recording
18. ct was purchased For Customer support for products purchased directly from AudioCodes contact support audiocodes com Abbreviations and Terminology Each abbreviation unless widely used is spelled out in full when first used Only industry standard terms are used throughout this manual Version 5 6 5 September 2008 ca AudioCodes MediaPack Series Related Documentation Document Manual Name LTRT 523xx where xx denotes Product Reference Manual the document version LTRT 665xx CPE SIP Configuration Guide for IP Voice Mail LTRT 654xx MP 11x amp MP 124 SIP User s Manual LTRT 598xx MP 11x amp MP 124 SIP MGCP Installation Guide Throughout this manual the terms MediaPack or device refer to the MP 124 MP 118 MP 114 and MP 112 VoIP gateways Throughout this manual the term MP 11x refers to the MP 118 MP 114 and MP 112 MediaPack series VoIP gateways SIP Release Notes 6 Document LTRT 65611 SIP Release Notes 1 What s New in Release 5 6 1 What s New in Release 5 6 This document uses a one row table convention to indicate the products for which each feature is applicable The products that don t support the feature are shaded grayed In the example below the feature would be applicable only to MP 11x FXS 11 Supported Hardware Platforms 1 1 1 New Products Introduced in this Release The following new product has been introduced in this release m MP 124 Rev D for DC power 1 1 2 Support of the Exis
19. dB steps The default is 0 i e no gain Defines the maximum size in bytes of a T 38 buffer supported by the device This value is included in the outgoing SDP when T 38 is used for fax relay over IP The valid range is 100 to 1 024 The default value is 1 024 22 Document LTRT 65611 SIP Release Notes Parameter FaxModemNTEMode 1 What s New in Release 5 6 Description Determines whether the device sends RFC 2833 ANS ANSAM events upon detection of fax and or modem answer tones i e CED tone 0 Disabled default 1 Enabled Note This parameter is applicable only when the fax or modem transport type is set to bypass or Transparent with Events 1 7 Modified Parameters The table below lists parameters from the previous release that have been modified for Release 5 6 The parameters enclosed in square brackets depict the ini file parameter the other parameters depict the parameters in the Embedded Web Server Table 1 2 Release 5 6 Modified Web ini File Parameters Parameter IPProfile Version 5 6 Description Modification Addition of DisconnectOnBrokenConnection This ini file table parameter configures the IP profiles table The format of this parameter is as follows IPProfile FORMAT IPProfile Index IPProfile_ProfileName IPProfile_IpPreference IPProfile CodersGroupID IPProfile_IsFaxUsed PProfile_JitterBufMinDelay IPProfile_JitterBufOptFactor PProfile_IPDi
20. der definition is disregarded e When an invalid rate is used for dynamic rate coders the coder definition is disregarded The device supports only symmetrical coders the same coder is used for transmit and receive though different ptime is supported The Transparent coder doesn t use DSP resources therefore the DSP functionality is off i e DTMF detection silence detection etc and a reset is needed before switching to a different coder Transcoding is not supported with coder frame sizes other than the default size refer to SampleBasedCodersRTPPacketinterval SIP Release Notes 44 Document LTRT 65611 SIP Release Notes 3 Known Constraints 26 27 28 Tables that use the improved ini file representation can t be burned to flash memory as Client Defaults It is highly recommended to use 100Base T switches Use of 10Base T LAN hubs should be avoided Static NAT is not supported for local IP calls 3 3 Web Constraints This release includes the following known Web constraints 1 10 11 12 13 Version 5 6 For MP 11x the Home page is not displayed correctly when the number of channels is reduced The window scrolling for the Home page sometimes does not function correctly when the window is resized There is no option to load an FXO Coefficient file to the device using the Auxiliary Files page If the Home button is clicked when the device Scenario mode is active
21. ds with an ACK and disconnects the call The device doesn t generate this response Upon receipt of this message and before a 200 OK has been received the device responds with an ACK and disconnects the call 40 Document LTRT 65611 SIP Release Notes 4xx Response 484 Address Incomplete 485 Ambiguous 486 Busy Here 487 Request Canceled 488 Not Acceptable 491 Request Pending Supported Yes Yes Yes Yes Yes Yes 2 Supported Features Comments The device doesn t generate this response Upon receipt of this message and before a 200 OK has been received the device responds with an ACK and disconnects the call The device doesn t generate this response Upon receipt of this message and before a 200 OK has been received the device responds with an ACK and disconnects the call The SIP device generates this response if the called party is off hook and the call cannot be presented as a call waiting call Upon receipt of this response the device notifies the User and generates a busy tone This response indicates that the initial request is terminated with a BYE or CANCEL request The device doesn t generate this response Upon receipt of this message and before a 200 OK has been received the device responds with an ACK and disconnects the call When acting as a UAS the device sent a re INVITE on an established session and is still in progress If it receives a re INVITE on the same dial
22. e 5 6 Obsolete Web ini File Parameters 0 ccccceeeececeeeeeeeeeeeeeeseeeeeeaeeeeneeenaes 29 Taole 2 BRE paigis i boi al arb going aera eee per eres rene AaS 34 Tea et SIP tastes assy cece esl ce cans andthe EE ms 34 Table 2 3 Sup period SIF Heater e s cccciscenticaciceena a teed ida 35 Tea a SDF Hedel ecaa ai a NEES 37 Table 2 5 Supported xx SIP Responso creccresecccsiierdesatue reatesecestansesecidierdaatears niniin in i ia 38 Table 2 6 Supported 2xx SIP RESPONSES crcessssec ae EG HAIR SIBH ontanned inn iniiae ai 38 Table 2 7 Supported 3xx SIP RESPONSES ceseceseccesiesserchetieerasteseccctansed ce cidinedasteuais ninii inni niia ig AA 39 Table 2 8 Supported dxx SIP RESPONSES csesgaN8 00606 HEA SHAGHAS NIGH SAISHSSSA RSS SAS EA EA SE SA 9640 Eii 39 Table 2 9 Supported 5xx SIP Responses isc iicascsssicadsissccdaedssiiddscssriassieoistsaioondisesscumiertaadecsrmmesneee 41 Table 2 10 Supported 6xx SIP Responses ssccesnneaan nee GAR AEA SIARSS INA N 6900360 06 6633606 8643156069043384 60603886 Ea 42 SIP Release Notes 4 Document LTRT 65611 SIP Release Notes Notices Notice This document describes the release of the AudioCodes MP 11x and MP 124 MediaPack Series of Voice over IP VoIP media gateways Information contained in this document is believed to be accurate and reliable at the time of printing However due to ongoing product improvements and revisions AudioCodes cannot guarantee accuracy of printed material a
23. e attempts to route the call to the Proxy If routing to the Proxy also fails the Redirect Transfer request is rejected When this parameter is set to 2 the XferPrefix parameter can be used to define different routing rules for redirected calls This parameter is disregarded if the parameter AlwaysSendToProxy is set to 1 Determines the size in bytes of the Master Key Identifier MKI in SRTP Tx packets The range is 0 to 4 The default value is 0 On a secured RTP session this parameter determines whether to enable Authentication on transmitted RTP packets 0 Enable default 1 Disable 21 September 2008 ca AudioCodes Parameter Disable Encryption On Transmitted RTP Packets RTPEncryptionDisableTx Disable Encryption On Transmitted RTCP Packets RTCPEncryptionDisableTx Alt Routing Tel to IP Keep Alive Time AltRoutingTel2IPKeepAliveTime RemoveToTaginFailureResponse FXOAutoDialPlayBusyTone SSHAdminKey SSHRequirePublicKey FaxBypassOutputGain ModemBypassOutputGain T38FaxMaxBufferSize SIP Release Notes MediaPack Series Description On a secured RTP session this parameter determines whether to enable Encryption on transmitted RTP packets 0 Enable default 1 Disable On a secured RTP session this parameter determines whether to enable Encryption on transmitted RTCP packets 0 Enable default 1 Disable Defines the time interval in sec
24. e responds with an ACK and disconnects the call The device doesn t generate this response Upon receipt of this message and before a 200 OK has been received the device responds with an ACK and disconnects the call If the device receives a 415 Unsupported Media response it notifies the User with a Reorder Tone The device generates this response in case of SDP mismatch The device doesn t generate this response Upon receipt of this message and before a 200 OK has been received the device responds with an ACK and disconnects the call The device does not generate this response On reception of this message the device uses the value received in the Min Expires header as the registration time If the device receives a 433 Anonymity Disallowed it sends a DISCONNECT message to the PSTN with a cause value of 21 Call Rejected In addition the device can be configured using the Release Reason Mapping to generate a 433 response when any cause is received from the PSTN side If the device receives a 480 Temporarily Unavailable response it notifies the User with a Reorder Tone This response is issued if there is no response from remote The device doesn t generate this response Upon receipt of this message and before a 200 OK has been received the device responds with an ACK and disconnects the call The device doesn t generate this response Upon receipt of this message and before a 200 OK has been received the device respon
25. el2lp Notes RemoveFromLeft RemoveFromRight Prefix2Add Suffix2Add LeaveFromRight NumberType NumberPlan and IsPresentationRestricted are applied if the called and calling numbers match the DestinationPrefix and SourcePrefix conditions The manipulation rules are executed in the following order RemoveFromLeft RemoveFromRight LeaveFromRight Prefix2Add and Suffix2Add Parameters can be skipped by using two dollar signs IsPresentationRestricted is set to Restricted only if Asserted Identity Mode is set to P Asserted Number Plan and Type can optionally be used in Remote Party ID RPID header by using the EnableRPIHeader SIP Release Notes 28 Document LTRT 65611 SIP Release Notes 1 What s New in Release 5 6 1 8 Obsolete Parameters The table below lists parameters from the previous release that are now obsolete Table 1 3 Release 5 6 Obsolete Web ini File Parameters Parameter SendinviteToProxy OfferUnencryptedSRTCP TestMode SASShortNumberLength Version 5 6 Description This parameter is obsolete instead use SIPReRoutingMode This parameter is obsolete instead use RTCPEncryptionDisableTx This parameter is now obsolete This parameter is obsolete instead use SASRegistrationManipulation 29 September 2008 7a 4 id AudioCodes MediaPack Series Reader s Notes SIP Release Notes 30 Document LTRT 65611 SIP Release Notes 2 Supported Feature
26. en establishing TLS connections 0 Disable default 1 Verify Subject Name only when acting as a server for the TLS connection 2 Verify Subject Name when acting as a server or client for the TLS connection When a remote certificate is received and this parameter is not disabled the SubjectAltName value is compared with the list of available Proxies If a match is found for any of the configured Proxies the TLS connection is established The comparison is performed if the SubjectAltName is either a DNS name DNSName or an IP address If no match is found and the SubjectAltName is marked as critical the TLS connection is not established If DNSName is used the certificate can also use wildcards to replace parts of the domain name If the SubjectAltName is not marked as critical and there is no match the CN value of the SubjectName field is compared with the parameter TLSRemoteSubjectName If a match is found the connection is established Otherwise the connection is terminated TLSRemoteSubjectName Modification Support for the asterisk wildcard Defines the Subject Name that is compared with the name defined in the remote side certificate when establishing TLS connections If the SubjectAltName of the received certificate is not equal to any of the defined Proxies Host names IP addresses and is not marked as critical the Common Name CN of the Subject field is compared
27. er Note The call flow depicted above is applicable only when SAS is in Emergency Mode Relevant parameter SASEnableENUM 4 Manipulation of AoR in Incoming REGISTER Requests for SAS Applications MP 124 MP 11x FXS FXO The SAS application now supports an option to manipulate the User Part of an incoming REGISTER request Address Of Record AoR before saving it to the registered users database The manipulation can include removing a certain number of digits from the right end of the number i e suffix or alternatively to keep only a certain number of digits from the right end of the number referred to as short numbering The registered database contains the AoR before and after the manipulation Note The parameter SASShortNumberLength is now obsolete Relevant parameter SASRegistrationManipulation Version 5 6 17 September 2008 7a id AudioCodes MediaPack Series 1 4 Web New Features The device supports the following new Web interface feature 1 Improved Interface for Manipulation Tables MP 124 MP 11x FXS FXO The Manipulation tables Tel to IP and IP to Tel source and destination numbers GUI interface has been enhanced to allow adding deleting and modifying of individual row entries 1 5 SNMP New Features The device supports the following new SNMP features 1 SNMP Alarm Raised when in SAS Emergency Mode MP 124 MP 11x FXS FXO The SAS application now generates an SNMP alarm when switching
28. er is not found the request is forwarded to the next redundant SAS defined in the Redundant SAS Proxy Set If that SAS Proxy IP appears in the Via header of the request it is not forwarded so that loops are prevented in the request s course If no such redundant SAS exists the SAS sends the 19 September 2008 ca AudioCodes Parameter SASSurvivabilityMode SASBindingMode SASEnableENUM SASRegistrationManipulation SIP Release Notes MediaPack Series Description request to its default gateway configured by the parameter SASDefaultGatewayIP The valid range is 1 to 5 The default value is 1 i e no redundant Proxy Set Determines the Survivability mode used by the SAS application 0 Standard All incoming INVITE and REGISTER requests are forwarded to the defined Proxy list in SASProxySet in Normal mode and handled by the SAS application in Emergency mode default 1 Always Emergency The SAS application does not use Keep Alive messages towards the SASProxySet and instead always operates in Emergency mode as if no Proxy in the SASProxySet is available 2 Ignore REGISTER Use regular SAS Normal Emergency logic same as option 0 but when in Normal mode incoming REGISTER requests are ignored Determines the SAS application database binding mode 0 URI If the incoming AoR in the INVITE requests is using a tel URI or user phone is defined the binding is performed
29. es the same way that it processes the 100 Trying response 182 Queued Yes The SIP device generates this response in Call Waiting service When the SIP device receives a 182 response it plays a special waiting Ringback tone to the telephone side 183 Session Yes The SIP device generates this response if the Early Media Progress feature is enabled and if the device plays a Ringback tone to IP 2 2 5 2 2xx Response Successful Responses Table 2 6 Supported 2xx SIP Responses 2xx Response Supported Comments 200 OK Yes 202 Accepted Yes SIP Release Notes 38 Document LTRT 65611 SIP Release Notes 2 Supported Features 2 2 5 3 3xx Response Redirection Responses 3xx Response 300 301 302 305 380 Multiple Choice Moved Permanently Moved Temporarily Use Proxy Alternate Service Table 2 7 Supported 3xx SIP Responses Supported Yes Yes Yes Yes Yes Comments The device responds with an ACK and then resends the request to the first new address in the contact list The device responds with an ACK and then resends the request to the new address The SIP device generates this response when call forward is used to redirect the call to another destination If such a response is received the calling device initiates an INVITE message to the new destination The device responds with an ACK and then resends the request to a new address The device responds with an ACK and
30. f a Busy tone Relevant parameters PProfile TelProfile 7 Maximum Row Entries Increased in Destination Number Manipulation Tables MP 124 MP 11x FXS FXO The maximum number of row entries in the Destination Number Manipulation tables has been increased to 100 Relevant parameters NumberMapTel2IP NumberMapIP2Tel 8 Maximum Row Entries Increased in Tel to IP Source Number Manipulation Table MP 124 MP 11x FXS FXO The maximum number of row entries in the Tel to IP Source Number Manipulation table has been increased to 120 rows Relevant parameter SourceNumberMapTel2IP 9 Maximum Row Entries Increased in the Internal DNS Table MP 124 MP 11x FXS FXO The maximum number of row entries in the Internal DNS table has been increased to 20 In addition each row now supports up to four different IP addresses Relevant parameter DNSZ2IP SIP Release Notes 10 Document LTRT 65611 SIP Release Notes 1 What s New in Release 5 6 10 Fields Source Trunk Group and Source IP Group Added to Manipulation Tables MP 124 MP 11x FXS FXO 11 New columns were added to the Destination and Source Number Tel to IP Manipulation tables to allow manipulation according to Source Trunk Group or Source IP Group Relevant parameters NumberMapTel2IP SourceNumberMapTel2P SRTP Enhancements MP 124 MP 11x FXS FXO The device now supports the following enhancements when using Secure Real time Transport Protocol SRTP e
31. ffServ IPProfile_SigIPDiffServ N A IPProfile_RTPRedundancyDepth IPProfile RemoteBaseUDPPort IPProfiie CNGmode IPProfile_VxxTransportType IPProfile_ NSEMode N A IPProfile_PlayRBTone2IP IPProfile_EnableEarlyMedia IPProfile_ProgressIndicator2IP IPProfile_ EnableEchoCanceller IPProfile_MediaSecurityBehaviour PProfile_CallLimit IPProfile DisconnectOnBrokenConnection IPProfile For example IPProfile IPProfile_1 name1 2 1 0 10 13 15 44 1 1 6000 0 2 0 0 0 1 0 1 0 1 1 IPProfile_2 name2 5 5 5 55 5 55 5 5 5 5 40 IPProfile Notes This parameter can appear up to 9 times i e indices 1 9 Indicates common parameters used in both IP and Tel profiles pPreference determines the priority of the Profile 1 to 20 23 September 2008 7a 4 id AudioCodes MediaPack Series Parameter Description where 20 is the highest preference If both IP and Tel profiles apply to the same call the coders and other common parameters indicated with an asterisk of the preferred Profile are applied to that call If the Tel and IP profiles are identical the Tel Profile parameters are applied Two adjacent dollar signs indicate that the parameter s default value is used PProfile can be used in the Tel to IP Routing and IP to Hunt Group Routing tables Prefix and PSTNPrefix parameters The Profile Name assigned to a Profile
32. fix SourceNumberMapTel2lp_SourceAddress SourceNumberMapTel2 p_NumberType 27 September 2008 7a 4 id AudioCodes MediaPack Series Parameter Description SourceNumberMapTel2 lp_NumberPlan SourceNumberMapTel2lp_RemoveFromLeft SourceNumberMapTel2 lp_RemoveFromRight SourceNumberMapTel2lp_LeaveFromRight SourceNumberMapTel2lp_Prefix2Add SourceNumberMapTel2lp_Suffix2Add SourceNumberMapTel2lp_IsPresentationRestricted NumberMapTel2lp_SrcTrunkGroupID NumberMapTel2lp_SrclPGroupID SourceNumberMapTel2lp Where DestinationPrefix Destination number prefix SourcePrefix Source number prefix SourceAddress Source IP address obtained from the Request URI in the INVITE message NumberType Number Type used in RPID header NumberPlan Number Plan used in RPID header RemoveFromLeft Number of stripped digits from the left RemoveFromRight Number of stripped digits from the right LeaveFromRight Number of remaining digits from the right Prefix2Add String to add as prefix Suffix2Add String to add as suffix IsPresentationRestricted Calling number presentation 0 to allow presentation 1 to restrict presentation SrcTrunkGroupID Source Trunk Group ID SrcIPGrouplD Source IP Group ID For example SourceNumberMapTel2lp SourceNumberMapTel2lp 0 22 03 0 0 2 667 0 SourceNumberMapTel2Ip 0 10 10 255 255 3 0 5 100 255 SourceNumberMapT
33. fter the Date Published nor can it accept responsibility for errors or omissions Updates to this document and other documents can be viewed by registered customers at http www audiocodes com support Copyright 2008 AudioCodes Lid All rights reserved This document is subject to change without notice Date Published Sep 14 2008 Date Printed Sep 15 2008 When viewing this manual on CD Web site or on any other electronic copy all cross references are hyperlinked Click on the page or section numbers shown in blue to reach the individual cross referenced item directly To return back to the point from where you accessed the cross reference press the ALT and keys Trademarks AC logo Ardito AudioCoded AudioCodes AudioCodes logo CTI CTI Squared InTouch IPmedia Mediant MediaPack MP MLQ NetCoder Netrake Nuera Open Solutions Network OSN Stretto 3GX TrunkPack VoicePacketizer VolPerfect What s Inside Matters Your Gateway To VoIP are trademarks or registered trademarks of AudioCodes Limited All other products or trademarks are property of their respective owners WEEE EU Directive Pursuant to the WEEE EU Directive electronic and electrical waste must not be disposed of with unsorted waste Please contact your local recycling authority for disposal of this product Customer Support Customer technical support and service are provided by AudioCodes Distributors Partners and Resellers from whom the produ
34. g 5 0 This can result in an invalid configuration 3 1 3 2 Version 5 6 SIP Constraints This release includes the following known SIP constraints 1 Channel parameters such as voice DTMF gain and jitter buffer are collectively configured in the ini file per device not per call By using Profiles this limitation can be overcome The number of RTP payloads packed in a single G 729 packet M channel parameter is limited to 5 Gateway Constraints This release includes the following known gateway constraints 1 In certain cases when the Spanning Tree algorithm is enabled on the external Ethernet switch port that is connected to the device the external switch blocks all traffic from entering and leaving the device for some time after the device is reset This may result in the loss of important packets such as BootP and TFTP requests which in turn may cause a failure in device start up A possible workaround is to set the ini file parameter BootPRetries to 5 causing the device to issue 20 BootP requests for 60 seconds Another workaround is to disable the spanning tree on the port of the external switch that is connected to the device PPPoE is not supported NTT caller ID Type 2 constraints e The NTT standard describes the CallerID Type 2 generation as a sequence of an incoming call signal C and D DTMFs and FSK modulated Data Generation of the incoming call signal remains the responsibility of the appli
35. h time an alternate route is found a tone is played for a user defined duration Once the tone has finished playing the new SIP INVITE is generated toward the new destination Note Tone Type 25 must be defined in the Call Progress Tones CPT file Relevant parameter A tRoutingToneDuration 15 New Re Routing Options for Redirect Transfer Scenarios MP 124 MP 11x FXS FXO When a call initiated by the device is re directed i e a 3xx SIP response is received or transferred i e a SIP REFER request is received several re routing options can now be selected e Send INVITE messages directly to the URI according to the Refer To header in the REFER message or Contact header in the 3xx response e Send a new INVITE message to the Proxy e Use the Routing table to locate the destination and then send the new INVITE to this destination This feature can be applied per device or per IP Group Note This feature replaces the existing SendINVITEToProxy parameter Relevant parameters S PReroutingMode PGroup 16 Wildcards Support in TLS Certificates MP 124 MP 11x FXS FXO The device now supports the receipt of wildcards in X 509 Certificates when establishing TLS connections These wildcards can be part of the CN attribute of the Subject field or the DNSName attribute of the SubjectAltName field 17 Multiple Digits in Dialing Plan Notation MP 124 MP 11x FXS FXO Currently the dialing plan for destination source
36. he call is released Determines whether the device plays a Comfort Tone Tone Type 18 to the FXS FXO endpoint after a SIP INVITE is sent and before a 18x response is received 0 Disable default I Enable Defines the prefix that is added to the destination number received in the SIP Refer to header in IP to Tel calls This parameter is applicable for FXO Blind Transfer modes LineTransferMode 1 2 or 3 The valid range is a string of up to 9 characters The default is an empty string Determines the time period in milliseconds that the device plays a tone on each Alternative Routing attempt When the tone finishes playing a new SIP INVITE message is generated toward the new destination The tone played is the Call Forward Tone i e Tone Type 25 in the CPT file The valid range is 0 to 20 000 The default time is 0 i e no tone is played Determines the Proxy Set index number used in SAS Normal mode to forward REGISTER and INVITE requests from the users that are served by the SAS application The valid range is 0 to 5 The default value is 0 i e default Proxy Set Determines the Proxy Set index number used in SAS Emergency mode for fallback when the user is not found in the Registered Users database Each time a new SIP request arrives the SAS application checks whether the user exists in the registration database If the user is located in the database the request is sent to the user If the us
37. he device s Stand Alone Survivability SAS application to operate in different Survivability modes e Immediately operate in Emergency Mode by setting the parameter SASProxySet to 1 In this case the SAS application does not send keep alive messages to the configured Proxy server and handles the incoming REGISTER and INVITE messages according to the Emergency mode settings e Operate according to the regular Normal Emergency logic but while in Normal mode REGISTER requests are ignored thereby forcing registering endpoints to switch to the serving Proxy instead of the SAS application Relevant parameter SASSurvivabilityMode 2 Cascading SAS Servers MP 124 MP 11x FXS FXO The SAS application now supports cascading of several SAS servers Each SAS application can use a Proxy Set as a redundancy mechanism The SAS application uses the Proxy Keep Alive mechanism to verify the status of each IP address defined in the Proxy Set marks each server as Online or Offline Relevant parameter RedundantSASProxySet SIP Release Notes 14 Document LTRT 65611 SIP Release Notes 1 What s New in Release 5 6 Each time a new SIP request arrives the SAS application checks whether the user is listed in the registration database If the user is listed in the database the request is sent to the specific user If the user is not found the request is forwarded to the next redundant SAS defined in the Redundant SAS Proxy Set If this specific SA
38. index must enable users to identify it intuitively and easily TelProfile Modification Addition of EnableDIDWink IsTwoStageDial and DisconnectOnBusyTone parameters This ini file table parameter configures the Tel Profile Settings table The format of this parameter is as follows TelProfile FORMAT TelProfile_Index TelProfile_ProfileName TelProfile_TelPreference TelProfile_CodersGroupID TelProfile_IsFaxUsed TelProfile_JitterBufMinDelay TelProfile_JitterBufOptFactor TelProfile_IPDiffServ TelProfile_SigIPDiffServ TelProfile_DtmfVolume TelProfile_InputGain TelProfile_VoiceVolume TelProfile_EnableReversePolarity TelProfile_ EnableCurrentDisconnect TelProfile_EnableDigitDelivery TelProfile_EnableEC TelProfile_MW1IAnalog TelProfile_MW Display TelProfile_FlashHookPeriod TelProfile_EnableEarlyMedia TelProfile_ProgressIndicator2IP TelProfile_ TimeForReorderTone TelProfile_EnableDIDWink TelProfile_IlsTwoStageDial TelProfile_DisconnectOnBusyTone TelProfile Indicates common parameters used in both IP and Tel profiles TelPreference determines the priority of the Profile 1 to 20 where 20 is the highest preference If both IP and Tel profiles apply to the same call the coders and other common parameters indicated with an asterisk of the preferred Profile are applied to that call If the preference of the Tel and IP profiles is identical the Tel Profile parameters are applied F
39. is reduced when G 729 is used with ptime 120 and G 723 is used with ptime 150 Therefore using these ptimes is not recommended When loading an ini file using the Web interface the swwd messages appears 45 September 2008 7a W E EA AudioCodes MediaPack Series 3 4 SNMP Constraints This release includes the following known Simple Network Management Protocol SNMP constraints 1 10 11 12 13 14 15 SNMP traps are not received when configuring more than one SNMP v3 trap destination A single GET command to the inetCidrRoute Table may return a No Such Instance error while GET NEXT as in WALK functions correctly When configuring the acSysinterfaceTable using SNMP or the Web interface validation is only performed after device reset When enabling Telnet using SNMP a fail notification is displayed despite the operation being successful When defining or deleting SNMPv3 users the v3 trap user must not be the first or last to be defined If there are no non default v2c users this results in a loss of SNMP contact with the device In the ipCidrRouteTable new rows cannot be added and rows that were previously deleted using the Web interface cannot be deleted The SNMPv3 users table returns the line removed notice when adding a new row to an active row index After adding an empty line to the SNMPV3 table it s impossible to delete it or add new lines The default values created in
40. l flow Figure 1 1 Double Hold SIP Call Flow Endpoint C Endpoint A Endpoint B Endpoint D INVITE sendrecv 200 OK sendrecv I r I I I i lt Conversation gt I I I I I I I 1 I I I I l I l j I l I I INVITE Hold inactive 200 OK inactive gt I I I I I I INVITE Hold inactive 200 OK inactive Conversation gt 0 d3 94 Ieig 200 OK sendrecv INVITE Retrieve sendrecv Conversation gt I r I I I I I I I I I I INVITE sendrecv I l I I I I I I I I I INVITE Hold inactive L 200 OK inactive 200 OK inactive I 200 OK sendrecv J 200 OK inactive Conversation gt Version 5 6 9 September 2008 7a T tal AudioCodes MediaPack Series Notes e Call Transfer while in a Double Hold state placing the phone on hook disconnects both calls i e call transfer is not performed e Call Waiting now supported while on hold The endpoint hears the Call Waiting tone instead of the Held tone 6 Additional Parameters in IP and Tel Profile Pages MP 124 MP 11x FXS FXO The IP and Tel Profile pages provide additional parameters to perform the following e P Profile Enables or disables the Broken Connection mechanism e Tel Profile Enables or disables DID Wink Selects the Dialing Mode One Stage or Two Stage Enables or disables disconnection of the call upon detection o
41. og it returns a 491 response to the received INVITE When acting as a UAC If the device receives a 491 response to a re INVITE it starts a timer After the timer expires the UAC tries to send the re INVITE again 2 2 5 5 5xx Response Server Failure Responses Table 2 9 Supported 5xx SIP Responses 5xx Response 500 Internal Server Error 501 Not Implemented 502 Bad gateway 503 Service Unavailable 504 Gateway Timeout 505 Version Not Supported Version 5 6 Comments Upon receipt of any of these Responses the device releases the call sending an appropriate release cause to the PSTN side The device generates a 5xx response according to the PSTN release cause coming from the PSTN 41 September 2008 7a 4 tal AudioCodes MediaPack Series 2 2 5 6 6xx Response Global Responses 600 603 604 606 Table 2 10 Supported 6xx SIP Responses 6xx Response Comments Busy Everywhere Decline Upon receipt of any of these Responses the device releases the call sending an appropriate Does Not Exist Anywhere release cause to the PSTN side Not Acceptable SIP Release Notes 42 Document LTRT 65611 SIP Release Notes 3 Known Constraints 3 Known Constraints This section lists known constraints in Release 5 6 Note Due to the improved ini file format for tables it s not possible to load an ini file that was used by a device running software version 5 2 or later to a device using an earlier version e
42. onds between SIP OPTIONS Keep Alive messages used for the IP Connectivity application The valid range is 5 to 2 000 000 The default value is 60 Determines whether the device removes the to header tag from final SIP failure responses to INVITE transactions 0 Do not remove tag default 1 Remove tag Determines whether the FXO device plays a Busy Reorder tone to the TDM side if a Tel to IP call is rejected by a SIP error response 4xx 5xx or 6xx The FXO device seizes the line off hook if a SIP error response is received and plays a Busy Reorder tone to the TDM side for the duration defined by the parameter TimeForReorderTone After playing the tone the line is released on hook 0 Disable default 1 Enable Determines the RSA public key for strong authentication to logging in to the Secure Shell SSH interface if enabled The value should be a base64 encoded string The value can be a maximum length of 511 characters For additional information refer to the Product Reference Manual Enables or disables RSA public keys for SSH 0 RSA public keys are optional if a value is configured for the ini file parameter SSHAdminKey default I RSA public keys are mandatory Defines the fax bypass output gain control The range is 31 to 31 dB in 1 dB steps The default is 0 i e NO gain Defines the modem bypass output gain control The range is 31 dB to 31 dB in 1
43. or example TelProfile TelProfile 1 FaxProfile 1 1 1 40 13 22 33 0 0 0 1 0 0 0 0 0 0 TelProfile 2 ModemProfile 2 2 0 40 13 0 0 0 0 TelProfile Notes This parameter can appear up to 9 times i e indices 1 9 Two adjacent dollar signs indicate that the parameter s default value is used SIP Release Notes 24 Document LTRT 65611 SIP Release Notes Parameter DNS2IP First Call Ringback Tone ID FirstCallRBTId Version 5 6 1 What s New in Release 5 6 Description The TelProfile index can be used in the Endpoint Phone Number table TrunkGroup parameter The Profile Name assigned to a Profile index must enable users to identify it intuitively and easily Modification Maximum number of table entries increased from 10 to 20 Maximum IP addresses increased from 2 to 4 This ini file table parameter configures the internal DNS table for resolving host names to IP addresses Four different IP addresses in dotted decimal notation can be assigned to a host name The format of this parameter is as follows Dns2lp FORMAT Dns2Ip Index Dns2lp_DomainName Dns2lp_FirstlpAddress Dns2lp_SecondlpAddress Dns2lp_ThirdlpAddress Dns2lp_FourthlpAddress Dns2lp Where DomainName Host name FirstlpAddress SecondlpAddress ThirdlpAddress FourthlpAddress First second third and fourth IP addresses respectively
44. phone_number domain name such as 122556 myproxy com Note that the SIP URI host name can be configured differently per called number Supports RFC 4040 RTP payload format for a 64 kbit s transparent data Can negotiate coder from a list of given coders Supports negotiation of dynamic payload types Supports multiple ptime values per coder Supports RFC 3389 RTP Payload for Comfort Noise Supports RFC 3824 Using E 164 numbers with SIP ENUM Supports receipt and DNS resolution of FQDNs received in SDP Supports lt draft ietf sip gruu 09 gt Obtaining and Using Globally Routable User Agent UA URIs GRUU in SIP Responds to OPTIONS messages both outside a SIP dialog and in mid call Generates SIP OPTIONS messages as Proxy keep alive mechanism Publishes the total number of free Tel channels in a 200 OK response to an OPTIONS requests Support RFC 3310 HTTP Digest Authentication Using Authentication and Key Agreement AKA Supports recepit of a REFER method outside of a dialog Support RFC 4458 SIP URIs for Applications such as Voicemail and Interactive Voice Response IVR Support RFC 3608 SIP Extension Header Field for Service Route Discovery During Registration Support RFC 3911 The SIP Join Header Partial Support RFC 4730 A SIP Event Package for Key Press Stimulus KPML Partial Support RFC 3455 Private Header P Header Extensions to SIP for the 3rd Generation Partnership Project 8GPP Parti
45. play a Busy Reorder tone to the TDM line if a SIP error response 4xx 5xx or 6xx is received The FXO device seizes the line off hook if a SIP error response is received and plays a Busy Reorder tone to the TDM side for the duration defined by the TimeForReorderTone parameter After playing the tone the line is released on hook Relevant parameter FXOAutoDialPlayBusy Tone 3 Play Comfort Tone to FXS FXO Endpoints MP 124 MP 11x FXS FXO The device now supports the option to play a Comfort Tone to the FXS or FXO endpoint Typically immediately after dialing is complete a SIP INVITE message is sent and after a certain period of time a SIP 18x response is received During this time interval i e after sending the INVITE and before receiving a 18x the device plays a Comfort Tone to the endpoint Relevant parameter EnableComfortTone 4 Hold Timeout MP 124 MP 11x FXS FXO The device now supports the option to keep a call on hold for a user defined time before disconnecting the call If a hold request SIP Re INVITE is received from the IP side a timer is started Unless a Retrieve request is received once the timer expires the call is disconnected Relevant parameter HeldTimeout SIP Release Notes 8 Document LTRT 65611 SIP Release Notes 1 What s New in Release 5 6 5 Double Hold The device now supports a Double Hold scenario where a Held party can senda Hold Re INVITE to the other side Example cal
46. prefixes in the Routing and Manipulation tables support the following notations e n m represents a range of numbers e n m represents multiple numbers e 2 3 4 xxx pound sign at the end of a number represents the end of a number SIP Release Notes 12 Document LTRT 65611 SIP Release Notes 1 What s New in Release 5 6 18 In previous releases the n m format only supported single digit numbers From this release it is now possible to use multiple digits up to three digits such as 11 22 33 or 111 222 333 The Dialing Plan notation is applicable to all manipulation and routing tables Time Interval between SIP OPTIONS Messages for IP Connectivity MP 124 MP 11x FXS FXO 19 It is now possible to configure the time interval between SIP OPTIONS Keep Alive messages used for the IP Connectivity application Relevant parameter A tRoutingTel2IPKeepAlive Time Add Prefix for Blind Transfer MP 124 MP 11x FXS FXO 20 The device now supports the option to add a prefix to the number defined in the SIP Refer To header for FXO Blind Transfer modes LineTransferMode 1 2 or 3 Relevant parameter XferPrefixlP2Tel Increased PRT Buffer Size MP 124 MP 124 Version 5 6 21 The pre recorded tone PRT buffer size has been increased from 100 to 200 Kbytes Configurable T 38 Fax Maximum Buffer Value in SDP MP 11x FXS FXO The device now supports specifying the maximum T 38 buffer size s
47. s 2 Supported Features 2 1 SIP Features 2 1 1 Supported SIP Features The device supports the following main SIP features Version 5 6 Reliable User Datagram Protocol UDP transport with retransmissions Transmission Control Protocol TCP Transport layer SIPS using TLS T 38 real time Fax using SIP Note If the remote side includes the fax maximum rate parameter in the SDP body of the INVITE message the device returns the same rate in the response SDP Operates with Proxy or without Proxy using an internal routing table Fallback to internal routing table if Proxy is not responding Supports up to 15 Proxy servers If the primary Proxy fails the device automatically switches to a redundant Proxy Supports domain name resolving using DNS NAPTR and SRV records for Proxy Registrar and domain names that appear in the Contact and Record Route headers Supports Load Balancing over Proxy servers using Round Robin or Random Weights Proxy or Registrar Registration such as REGISTER sip servername SIP 2 0 VIA SIP 2 0 UDP 212 179 22 229 branch z9hG4bRaC7AU2 34 From lt sip GWRegistrationName sipgatewayname gt tag 1c29347 To lt sip GWRegistrationName sipgatewayname gt Call ID 10453 212 179 22 229 Seq 1 REGISTER Expires 3600 Contact sip GWRegistrationName 212 179 22 229 Content Length 0 The servername string is defined according to the following rules e The servername i
48. s are allowed simultaneously Flash burning control for specific files BurnCallProgressToneFile is no longer supported The new SaveConfiguration parameter must be used instead VLAN Pass Through mode is not supported 10Base T Half Duplex is not supported only 10 100Base T Full Duplex and 100Base T Half Duplex are supported When using a sample interval of 10 or 5 msec the channel capacity may be reduced When using SRTP channel capacity is reduced Contact AudioCodes for more details When using SRTP the number of basic codec frames per RTP packet cannot be greater than one In addition the RTP Redundancy RFC 2198 feature cannot be activated The DJBufOptFactor parameter cannot be set to 13 if the channel is configured to operate with Silence Compression enabled When using m factor values greater than 8 you must set jitter buffer optimization to 13 to cancel any jitter optimization and avoid under running condition Date and time should be set after each device reset unless Network Time Protocol NTP is used The Syslog CDR Date and Time fields are left empty if the device s Date and Time are not set and NTP is not used Daylight Savings Time is not supported The following constraints apply when defining coders via the ini file e Coder names are case sensitive e Don t use obsolete coder names e g g729_ AnnexB g7231r53 with the improved coder interface e When an invalid packetization time is used the co
49. s equal to RegistrarName if configured The RegistrarName can be any string e Otherwise the servername is equal to RegistrarIP either FQDN or numerical IP address if configured e Otherwise the servername is equal to ProxyName if configured The ProxyName can be any string e Otherwise the servername is equal to ProxyIP either FQDN or numerical IP address The parameter GWRegistrationName can be any string This parameter is used only if registration is Per Gateway If the parameter is not defined the parameter UserName is used instead If the registration is per endpoint the endpoint phone number is used 31 September 2008 1 L tal AudioCodes MediaPack Series The sipgatewayname parameter defined in the ini file or set from the Web browser can be any string Some Proxy servers require that the sipgatewayname in REGISTER messages is set equal to the Registrar Proxy IP address or to the Registrar Proxy domain name The sipgatewayname parameter can be overwritten by the TrunkGroupSettings GatewayName value if the TrunkGroupSettings_RegistrationMode is set to Per Endpoint REGISTER messages are sent to the Registrar s IP address if configured or to the Proxy s IP address A single message is sent once per device or messages are sent per channel according to the parameter AuthenticationMode There is also an option to configure registration mode per Trunk Group using the TrunkGroupSettings ta
50. se 5 6 1 The Web User Accounts page does not support Scenario mode v This constraint is now supported 2 MP 118 and MP 124 When clicking the Uplink icon on the Home page the Ethernet Port Information page that opens sometimes displays incorrect Ethernet port information To correctly view this information navigate to Status and Diagnostics gt Ethernet Port Information v This constraint is now fixed 3 Screen resolution 1152 x 864 is not supported y This constraint is now supported 4 On the IP Settings page when selecting a multiple or dual value from the IP Networking Mode field the DHCP field is incorrectly enabled v This constraint is now fixed Version 5 6 47 September 2008 7a 4 id AudioCodes MediaPack Series Reader s Notes SIP Release Notes 48 Document LTRT 65611 SIP Release Notes 5 Earlier Releases 5 Earlier Releases Details of previous releases can be found in the Release Notes of Version 5 4 published by AudioCodes on May 20 2008 Version 5 6 49 September 2008 Ces AudioCodes CPE amp Access Analog Gateways SIP MediaPack MP 124 amp MP 11x Release Notes Version 5 6 www audiocodes com lar Cad AudioCodes
51. ting Hardware Platforms The following existing hardware platforms are supported in this release m MP 11x combined FXS FXO devices e MP 114 FXS FXO providing 2 FXS ports and 2 FXO ports e MP 118 FXS FXO providing 4 FXS ports and 4 FXO ports m MP 11x FXO devices e MP 118 FXO providing 8 analog FXO interfaces e MP 114 FXO providing 4 analog FXO interfaces m MP 11x FXS devices e MP 118 FXS providing 8 analog FXS interfaces e MP 114 FXS providing 4 analog FXS interfaces e MP 112 FXS providing 2 analog FXS interfaces m MP 124 FXS providing 24 analog FXS interfaces 113 Hardware Platforms No Longer Supported Not applicable Version 5 6 ri September 2008 7a c tal AudioCodes MediaPack Series 1 2 General Gateway New Features The device supports the following new gateway features 1 Immediate Release of Tel to IP Call when Device Receives 401 407 Response MP 124 MP 11x FXS FXO If the device s default password has never been modified and an Authentication Required SIP response 401 407 is received the call is immediately released and a SIP Re INVITE message is not sent 2 Play Busy Tone to IP Upon Call Failure FXO In previous releases when the FXO device operated in Automatic Dialing mode there was no method to inform the caller that the Tel to IP call failed The reason was that the FXO device does not seize the line until a SIP 200 OK response is received A new option has been added which allows the device to
52. upported by the device This value is included in the outgoing Session Description Protocol SDP Relevant parameter T38FaxMaxBufferSize 22 RSA Keys in SSH MP 11x FXS FXO The device s internal Secure Shell SSH server now supports RSA public keys By default SSH uses the same user name and password as the Telnet and Web servers In addition SSH supports 1024 bit RSA public keys which provide carrier grade security The device can now be configured with an administrator RSA key as a means of strengthening authentication For information on implementing public keys for SSH refer to the Product Reference Manual Relevant parameters SSHAdminkey SSHRequirePublicKey 13 September 2008 7a c tal AudioCodes MediaPack Series 23 Fax Modem Bypass Output Gain Configuration MP 124 MP 11x FXS FXO It is now possible to determine fax and or modem bypass output gain values by fine tuning the level of appropriate output signals in bypass VBD mode Relevant parameters FaxBypassOutputGain ModemBypassOutputGain 24 Sends RFC 2833 ANS ANSam Events Upon Fax Modem Answer Tones MP 124 MP 11x FXS FXO The device can now be configured to send RFC 2833 ANS ANSam events upon detection of fax and or modem answer tones i e CED tone Relevant parameter FaxModemNTEMode 1 3 SIP New Features The device supports the following new SIP features 1 Different SAS Modes MP 124 MP 11x FXS FXO It is now possible to configure t
53. with this value If not equal the TLS connection is not established If the CN uses a domain name the certificate can also use wildcards to replace parts of the domain name The valid range is a string of up to 49 characters Note This parameter is applicable only if the parameter PeerHostNameVerificationMode is set to 1 or 2 NumberMapTel2IP Modification New parameters for Source Trunk Group and Source IP Group This ini file table parameter manipulates manipulates the destination number of Tel to IP calls The format of this parameter is as follows NumberMapTel2lp FORMAT NumberMapTel2Ip_Index NumberMapTel2lp_DestinationPrefix NumberMapTel2lp_SourcePrefix NumberMapTel2lp_ SourceAddress NumberMapTel2lp_NumberType NumberMapTel2lp_NumberPlan NumberMapTel2lp_RemoveFromLeft NumberMapTel2lp_RemoveFromRight NumberMapTel2lp_LeaveFromRight NumberMapTel2lp_Prefix2Add NumberMapTel2lp_Suffix2Add SIP Release Notes 26 Document LTRT 65611 SIP Release Notes Parameter SourceNumberMapTel2IP Version 5 6 1 What s New in Release 5 6 Description NumberMapTel2lp_IsPresentationRestricted NumberMapTel2lp_SreTrunkGroupID NumberMapTel2lp_ SrcIPGrouplD NumberMapTel2lp Where DestinationPrefix Destination number prefix SourcePrefix Source number prefix SourceAddress N A NumberType Number Type used in RPID header NumberPlan Number Type used in RPID header Remove
54. xPacketMkISize Disable Authentication On Transmitted RTP Packets RTPAuthenticationDisableTx Version 5 6 1 What s New in Release 5 6 Description LeaveFromRight number of digits to keep from the right side If both RemoveFromRight and LeaveFromRight are defined the RemoveFromRight is applied first The registered database contains the AoR before and after the manipulation The range of both RemoveFromRight and LeaveFromRight is 0 to 30 Note This table can include only one index entry Determines the routing mode after a call redirection i e a 3xx SIP response is received or transfer i e a SIP REFER request is received 0 Standard INVITE messages that are generated as a result of Transfer or Redirect are sent directly to the URI according to the Refer To header in the REFER message or Contact header in the 3xx response default 1 Proxy Sends a new INVITE to the Proxy Note Applicable only if a Proxy server is used and the parameter AlwaysSendtoProxy is set to 0 2 Routing Table Uses the Routing table to locate the destination and then sends a new INVITE to this destination Notes When this parameter is set to 1 and the INVITE sent to the Proxy fails the device re routes the call according to the Standard mode 0 When this parameter is set to 2 and the INVITE fails the device re routes the call according to the Standard mode 0 If DNS resolution fails the devic

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